Communications system and method utilizing centralized signal processing

ABSTRACT

A communications system and method performs centralized signal processing on received audio signals. A plurality of terminals are coupled to a processing switch via links. The terminals can be, for example, dedicated speakerphones, desktop handsets, or personal computers with audio capabilities. The links can be wired or wireless, can carry analog or digital signals, and can be shared with other users in a network. The switch receives the audio data from the terminals, processes the data according to desired acoustical procedures, creates one or more output mixes, and provides the output mixes to the appropriate terminals. The operation of the processing switch is controlled by a communications support module (CSM) which can receive, process, and send data to/from multiple terminals simultaneously. The CSM receives audio signals from the terminals. The CSM uses stored room models holding room model information including data and/or filters representing the acoustic properties of the terminal and/or the environment surrounding the terminal to produce the audio signals. Signal processing (SP) modules provide a pool of SP power from which the CSM can draw to process audio signals received from or being sent to the terminals. The CSM uses the SP modules to perform signal processing including acoustic echo cancellation, automatic gain control, noise reduction. The CSM also uses a mixing module to perform signal mixing.

CROSS-REFERENCE TO RELATED APPLICATION

This application is a continuation of U.S. Ser. No. 09/660,205, filed onSep. 12, 2000, entitled COMMUNICATIONS SYSTEM AND METHOD UTILIZINGCENTRALIZED SIGNAL PROCESSING, by inventors James H. Parry and PeterHsiang, currently pending [Attorney Docket No. 38715-P002US].

FIELD OF INVENTION

This invention pertains in general to telephony and in particular toperforming centralized acoustic signal processing on audio signalsreceived from terminals engaged in communications sessions.

BACKGROUND OF INVENTION

Hands-free two-way audio communications systems, such as speakerphones,utilize both a microphone and a speaker. The microphone transmits speechand other sounds from the local terminal to remote terminals while thespeaker emits sounds received from remote terminals. In a typicalspeakerphone system, the speaker and microphone are located in closeproximity and sounds produced by the speaker are picked up by themicrophone. Without signal processing, therefore, a feedback loop iseasily created between the speaker and microphone. This feedback cancause the speaker to emit an undesirable “howling” noise and cause theremote terminals to hear echoes.

One simple technique for eliminating feedback is to provide half-duplexswitching where only the microphone or the speaker is active at anygiven instant. In a typical half-duplex system, the speaker is activeuntil a sound is detected at the microphone. Then, the speaker becomesinactive and the microphone becomes active for the duration of thesound. Half-duplex systems have many inherent problems, not the least ofwhich is that a slight noise may unintentionally cause the speaker tocut out. As a result, it is often difficult to conduct a normalconversation using a speakerphone with a half-duplex switching system.

More sophisticated audio communications systems use complicated adaptivetechniques to reduce background noises as well as to regulate gain inthe audio channel and eliminate feedback. These techniques identifyselected acoustical situations, such as “doubletalk” or “voice notpresent,” and use these identifications to control the rate ofadaptation of the signal conditioning methods. Adaptive acoustic echocancellation (AEC), for example, is performed at the speakerphone andutilizes a sample-by-sample copy of the signal going to the speaker asthe basis for an estimate of the echo returning through the microphone.The estimated echo is subtracted on a sample-by-sample basis in anattempt to separate out only that portion of the microphone signal dueto sounds coming from sources other than the speaker.

Other signal processing techniques may also be performed at the terminalto improve the quality of the audio signal. For example, frequencyshifting is sometimes used to further attenuate loop gain at aparticular frequency and thus avoid howling. In addition, a noisereduction algorithm can be used to estimate a frequency dependentprofile of the noise floor and attenuate sounds which are temporarilynear or below that noise floor. A voice-gated automatic gain control(AGC) can also be used to isolate times during which local speech isthought to be present and then adjust the signal gain so that the speechis near a predetermined level when considered on average.

These solutions can work reasonably well, but the software and hardwarefor implementing these solutions is integrated into the speakerphone.Thus, the software and hardware must be replicated for each speakerphoneand the total cost of the solution depends upon the number ofspeakerphones in existence. In addition, each speakerphone must bedesigned and built to some pre-selected level of quality and can not beeasily tuned to a particular use or environment. Similarly, it is noteasy to change or upgrade the solution implemented by the speakerphone.

Therefore, there is a need in the art for a solution that provideseffective signal processing to an audio communications system but doesnot have costs that scale with the number of speakerphones. Preferably,this solution would also allow the signal processing performed for eachspeakerphone to be easily tuned or upgraded for a particular use orenvironment.

DISCLOSURE OF INVENTION

The above needs are met by an audio communications system and methodthat performs centralized digital signal processing. Since there is noneed to provide digital signal processors in the terminals, the cost ofthe terminals is reduced. Moreover, the present invention allows easytailoring and upgrading of the capabilities of the communicationssystem.

In an embodiment of the present invention, a plurality of terminals arecoupled to a processing switch. The terminals can be, for example,dedicated speakerphones, desktop handsets, or personal computers withaudio capabilities. The terminals can be coupled to the switch via wiredand/or wireless links.

The processing switch provides support for audio communications. Thecentralized signal processing capabilities of the switch are provided bya communications support module (CSM). Under direction of the CSM, theswitch can receive, process, and send data to/from multiple terminalssimultaneously. In addition, the switch can support multiplesimultaneous communication sessions, where each session contains two ormore terminals engaged in communications.

In one embodiment, the CSM develops and stores room models having datasets representing the acoustic properties of the terminals and/or theterminals' environments. A room model can also hold state informationreflecting signals previously sent to the associated terminal.

The CSM is also supported by one or more signal processor (SP) modules.The SP modules provide a pool of processing resources from which the CSMcan draw to process audio signals received from or sent to theterminals. In one embodiment, the CSM can dynamically allocate anddeallocate SP resources in response to the overall system load on theprocessing switch or the characteristics of the particular audio signalsreceived from the terminals. Thus, if relatively few communicationssessions are being conducted on the switch, more processing power can beallocated to particular signals than if many sessions are occurringsimultaneously.

In operation, the processing switch receives audio signals from one ormore of the communications terminals. If necessary, the CSM decodes thereceived audio signals into a formal suitable for further processing.Then, the CSM processes the decoded signals according to desiredacoustical procedures.

The CSM uses the SP modules to perform signal processing on the receivedaudio signals. The types of signal processing available include acousticecho cancellation (AEC), automatic gain control, noise reduction, andsignal mixing. Preferably, the CSM uses the room models to determine theoptimal signal processing to perform on the audio signals. In oneembodiment, the CSM can dynamically allocate SP power to particularaudio signals in response to difficult room environments such as roomshaving long reverberation times or high noise levels.

The CSM is also preferably supported by a mixing module. The mixingmodule formulates an output mix for each of the terminals. In general,each terminal's output mix is comprised of a mix of the received audiosignals from the other terminals. If only one terminal is producing anaudio signal, then the formulated output mix merely contains theprocessed audio signal from the SP module and the terminal producing theaudio signal does not receive the output mix.

BRIEF DESCRIPTION OF DRAWINGS

FIG. 1 is a block diagram of a prior art audio communications system;

FIG. 2 is a high-level block diagram of an audio communications systemaccording to an embodiment of the present invention; and

FIG. 3 is a flowchart illustrating the operation of the communicationssupport module according to an embodiment of the present invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

FIG. 1 is a block diagram of a prior art switched audio communicationssystem 100. A plurality of terminals 110A-C are coupled to a centralswitch 112 via communications link 114A-C. Each terminal 110, of whichterminal 110A is representative, includes a microphone 116A and speaker118A. As uses herein, a speaker is any device that produces audiblemotion in response to an electrical signal and a microphone is anydevice that produces an electrical signal in response to audible motion.The communications links 114 carry sounds picked up by the microphone116 and to be played by the speaker 118 to/from the switch 112.

The terminal 110 contains a signal processor (SP) 120 with which theterminal performs acoustic echo cancellation (AEC). The AEC removes thespeaker 118 sounds that are picked up by the microphone 116. The switch112 performs switching and routing of audio signals by determining fromwhich terminal an audio signal is being received and to whichterminal(s) the audio signal should be sent.

FIG. 2 is a high-level block diagram of an audio communications system200 according to an embodiment of the present invention. A plurality ofterminals 210A-D are coupled to a processing switch 212 viacommunications links 214A-D. The terminal types can be heterogeneous. Inone embodiment, the terminals include: dedicated speakerphones, desktophandsets with or without speakerphone capabilities, cellular phones,desktop personal computer (PC) systems with audio capabilities, videoconferencing systems with audio capabilities, etc. Each terminal 210, ofwhich terminal 210A is representative, preferably includes a microphone216A and a speaker 218A. Unlike in the system of FIG. 1, the terminals210 need not, and preferably do not, contain a SP for performing AEC orother complex signal processing. However, the terminals 210 may have SPsfor performing other functions, such as analog-to-digital anddigital-to-analog-signal conversions and, optionally, dataencoding/decoding.

The communications links 214 carry audio data representative of soundspicked up by the microphone 216 and to be played by the speaker 218to/from the processing switch 212. The communications link 214 may bewired or wireless. Moreover, the links 214 may include dedicated privatelinks, shared links utilizing a publicly-accessible telephone network,and/or links using a public or private data communications network suchas the Internet. Data traveling over the links 214 may pass through oneor more switches or link types before reaching the processing switch 212or terminal 210, although a preferred embodiment of the presentinvention treats a link passing through multiple links and switches as asingle logical link. The data carried by the communications links 214can be digital and/or analog. If the data is digital, it is preferablytransmitted as a series of discrete data packets, such as Internetprotocol (IP) packets. In one embodiment, the digital data is encodedinto a compressed format.

The processing switch 212 switches and routes communications among theterminals 210 and provides signal processing as described herein. Theswitch 212 can be, for example, a private branch exchange (PBX) locatedat a business or other entity, a publicly-accessible switch operated bya telephone company or other entity providing audio communications, oran Internet server supporting Internet telephony. Thus, the term“processing switch” includes any device capable of providing theswitching and processing functionality attributed to the processingswitch 212 described herein.

In one embodiment, the processing switch 212 comprises a dedicatedcomputer system having one or more central processing units (CPUs), arandom access memory (RAM), read-only memory (ROM), a storage devicesuch as a hard drive, switching hardware and software, and otherhardware and software components for providing switch functionality asis known in the art.

Aggregations of machine-executable code, data, circuitry, and/or datastorage areas in the processing switch 212 for performing a specificpurpose or purposes are referred to as “modules.” Different modules mayshare common code, data, and/or circuitry. The modules include, forexample, modules for receiving and sending data, a digital to analogconverter (DAC) module, and an analog to digital converter (ADC) module.The switch 212 can perform multiple tasks simultaneously by allocating asubject of available modules, processors, processing time, or otherresources to particular tasks.

According to an embodiment of the present invention, the processingswitch 212 has a communications support module (CSM) 220 for supportingcommunications utilizing shared and centralized signal processing. Underdirection of the CSM 220, the switch 212 receives the data provided bythe communications links 214, processes the data using centralizedresources and modules, and provides the data to the appropriateterminals.

In one embodiment, the CSM 220 associates a room model 222 with eachterminal engaged in a communications session. The room model 222 holdsroom model information including data and/or filters representing theacoustic properties of the terminal 210 and/or the environmentsurrounding the terminal. For example, in one embodiment the room model222 holds data representing the reverberation characteristics of theroom in which the terminal 210 is located. The room model 222 can alsohold data representing an amount of background noise present at theterminal, an amount of automatic gain control (ACG) to be applied toaudio data received from the terminal, types of noise reduction to beapplied to signals received from or sent to the terminal, or any otherinformation useful for supporting signal processing to be performed ondata received from or sent to the associated terminal 210. Room modelinformation held in the room model 222 can also include stateinformation indicating signals previously sent to the associatedterminal 210. In one embodiment of the present invention, the resourcesutilized by the room model 222, such as memory and processing power, aredynamically allocated. For example, if a room is found to have a longreverberation time, a larger memory is allocated to the room model 222in order to store the sequencing information.

In one embodiment of the present invention, the room model 222 isadaptively developed while the terminal 210 is used in a communicationssession. This technique is preferred because the room model 222 maychange during the course of the communications session. For example, aparticipant may switch the terminal 210 from a handset to a speakerphonemidway through a conversation. Alternatively, the room model can begenerated by sending a series of test signals to the terminal atdifferent points during the session.

In one embodiment, room models 222 are persistently stored in theprocessing switch 212 and retrieved from storage each time an associatedterminal 210 is used in a communications session. For example, if theset of terminals interacting with the switch 212 is finite, known, andhas relatively constant acoustic properties, it may be more efficient tostore a persistent room model for each terminal 210. In contrast, if theswitch 212 is a server on the Internet and the set of terminals isunknown and practically infinite, it may be more efficient to generate anew room model each time a terminal establishes a connection with theswitch.

The CSM 220 is supported by one or more SP modules 224. The SP modules224 preferably operate in the digital domain, but may also provideanalog processing. Taken together, the SP modules 224 provide a pool ofprocessing resources from which the CSM can draw. During times when thesystem load is light (i.e., relatively few terminals are engaged inaudio communications), correspondingly greater processing resources areavailable for use with each supported terminal 210. Likewise, when thesystem load is heavy (i.e., a relatively large number of terminals 210are engaged in audio communications), the processing resources availablefor each terminal are diminished.

In a preferred embodiment of the present invention, the CSM 220 monitorsthe received audio signals and the processing needs of the switch 212and dynamically allocates and deallocates the processing resources ofthe SP modules 224 to the signals in order to provide maximum benefit tothe active terminals (i.e., terminals having microphones generatingaudio signals). In one embodiment, this monitoring is performed byanalyzing the digital packets received from the terminal 210 over thecommunications links 214. If the data in the packets forming the signalare determined to represent audio data generated by a person speaking,SP resources allocated to that signal. The amount of resources allocateddepends on the amount of SP resources available in the pool. If, on theother hand, the CSM 220 determines from the packets that no one isspeaking at the terminal, the CSM preferably digitally mutes the signaland allocates relatively few SP resources to it.

The CSM 220 is preferably supported by a mixing module 226. The mixingmodule 226 formulates an output mix for each of the terminals 210. Ingeneral, each terminal's output mix is comprised of a mix of thereceived audio signals from the other terminals. If only one terminal isproducing an audio signal, then the formulated output mix merelycontains the processed audio signal from the SP module 224 and no outputmix is sent to the terminal producing the audio signal. In oneembodiment, the mixing module 226 is merely a logical construct formedwhen the CMS 220 uses the SP modules 224 to perform mixing.

FIG. 3 is a flowchart illustrating the operation of the CSM 220 whenproviding real-time centralized signal processing according to oneembodiment of the present invention. For purposes of example, assumethere are N terminals engaged in a communications session (e.g., Nspeakerphones engaged in a conference call). N may be a subset of thetotal number of terminals coupled to the processing switch 212 and theremay be multiple communications sessions ongoing simultaneously. As isunderstood in the art, there are many different ways for the initialcommunications sessions to be established. For example, all of theparticipants can call into the switch 212 or one participant caninitiate the session by calling the other participants.

At step 310, the processing switch 212 receives audio signals from oneof the terminals 210 via the communications links 214. At any giventime, one or more of the terminals 210 can produce audio signals andsend the signals to the switch 212. Thus, the switch 212 cansimultaneously receive audio signals from multiple terminals. Audiosignals received by the switch 212 are made available to the CSM 220 andSP modules 224 in the switch.

Next, the CSM 220 uses the SP modules 224 to decode 312 the signalsreceived via the communications links 214, if necessary. As mentionedabove, the audio signals can be digitally encoded to compress thesignal, detect errors, and/or provide other benefits. The CSM 220decodes and/or decompresses the audio signals and stores the signals ina format suitable for performing further processing. Since differentterminals 210 may use different encoding/decoding methods, the CSM 220preferably simultaneously supports multiple encoding/decoding methods.

The CSM 220 uses the SP modules 224 to process 314 the decoded signalsaccording to desired acoustic procedures. The types of processing 314that can be performed on the decoded audio signals include: AEC, AGC,noise reduction, and mixing. Of course, this list is not exclusive andany type of desired processing can be performed on the signals. Thisprocessing may utilize the room models 222 and, in addition, may updatethe room models 222.

For adaptive AEC, the CSM 220 determines which terminals are active.Then, the CSM 220 uses the associated room model 222 to process thesignal from that terminal and remove the echo caused by the microphone216 picking up sounds from the speaker 218. In one embodiment, digitalsample values of the audio signals previously sent to the terminal 210are stored in the associated room model 222. The stored digital samplevalues are used to estimate the echo returning through the microphone216 of that terminal and the estimated echo is subtracted on asample-by-sample basis from the received audio signal. In oneembodiment, the outgoing packets bear sequencing information, such astime sequence tags, which is used to determine a correlated timesequence on the returned packets of audio information. In other words,the time sequence tags are used to maintain alignment between thesamples delivered to the terminal's speaker 218 and the samples receivedfrom the terminal's microphone 216.

For ACG, the CSM 220 isolates times during which local speech at aterminal 210 is thought to be present and then adjusts the signal gainso that the speech is near a predetermined level when considered onaverage. In one embodiment, the CSM 220 stores data in the room model222 indicating the amount of signal gain to apply to the associatedmicrophone 216.

For noise reduction, the CSM 220 estimates a frequency dependent profileof the noise floor and attenuates sounds which are temporarily near orbelow that noise floor. In one embodiment, the CSM 220 stores theestimated frequency dependent profile of a noise floor for a terminal inthe associated room model 222.

Once the signal processing is performed 314, the CSM 220 uses the mixingmodule 226 to formulate 316 output mixes of the signal. If only oneterminal 210 is producing an audio signal, and N is the number ofterminals 210 engaged in the communications session, then the mixingmodule 226 formulates N−1 output mixes, where each audio mix isassociated with a terminal other than the terminal from which the audiosignal was received. In this case, the output mix merely contains theprocessed signal produced by the SP modules 224. If more than oneterminal 210 is simultaneously producing an audio signal, then themixing module 226 formulates N output mixes—one mix for each terminal.In this latter case, each terminal's mix contains all of the receivedaudio signals except for any audio signal received from the terminal forwhich the mix was formulated. Since a preferred embodiment of thepresent invention digitally mutes those terminals at which the CSM 220determines no one is currently speaking, no processing power is requiredto mix signals from those terminals. In addition, the output mixes canoptionally be further processed according to the room model 222 for theassociated terminal 210.

Next, the CSM 220 uses the SP modules 224 to encode 318 each mix intothe appropriate format for its associated terminal 210. Since theencoding process can introduce noise or other artifacts into the signal,a preferred embodiment of the present invention decodes 320 each encodedmix and provides decoded mix samples to the associated room model 222.These samples become the sample values used to estimate the echoreturning through the microphone 216 of that terminal when performingAEC. Alternative embodiments of the present invention, however, usesamples made before the mix is encoded. The CSM 220 sends 322 each mixto its associated terminal via the communications links 214. Theterminal 210 plays the mix out of the terminal's speaker 218.

A preferred embodiment of the present invention supports easy upgradesof the processing switch 212 modules. The CSM 220, room models 222, andSP processing 224 modules are preferably software-upgradeable. Inaddition, one embodiment of the switch can be upgraded by replacing oradding modules. For example, the switch 212 can be upgraded by adding SPmodules. In this manner, the performance of the switch 212 can beimproved without altering the terminals 210 or otherwise incurringadditional cost.

The above description is included to illustrate the operation of thepreferred embodiments and is not meant to limit the scope of theinvention. The scope of the invention is to be limited only by thefollowing claims. From the above discussion, many variations will beapparent to one skilled in the relevant art that would yet beencompassed by the spirit and scope of the invention.

1. A method for supporting communication among a plurality of terminals,the method comprising steps of: receiving an audio signal at acentralized switch to result in a received signal, wherein thecentralized switch receives the audio signal from a terminal over acommunications medium, wherein the terminal has a microphone locatedproximate to not more than one active speaker mechanism, wherein theterminal lacks mechanisms for removing an audio feedback portion of theaudio signal, wherein the audio feedback portion is caused by themicrophone reacting to sound energy from the not more than one activespeaker mechanism; processing by a signal processor the received audiosignal to remove the audio feedback portion resulting in a firstfiltered signal; mixing the first filtered signal with a second filteredsignal to result in a mixed signal; and transmitting the mixed signal tothe terminal over the communications medium.
 2. The method of claim 1,wherein the communications medium is wireless.
 3. The method of claim 1further comprising the step of: extracting room model information fromthe received signal, wherein the room model information containscharacteristics of the acoustic qualities surrounding the terminal. 4.The method of claim 1, wherein the signal processor performs acousticecho cancellation on the received signal.
 5. The method of claim 1,wherein the signal processor performs automatic gain control on thereceived signal.
 6. The method of claim 1, wherein the signal processorperforms noise reduction on the received signal.
 7. The method of claim1 further comprising the steps of: extracting room data from thereceived signal, where in the room data contains characteristics ofacoustic properties surrounding the terminal; storing portions of theroom data in a first room model.
 8. The method of claim 7, wherein theroom model is adaptively developed and updated while the terminal isengaged in a communications session.
 9. The method of claim 7 furthercomprising the steps of: identifying a second terminal engaged in acommunications session to result in an identified terminal; andretrieving a second room model for the identified terminal.
 10. Themethod of claim 1, the method further comprising the steps of:determining an amount of processing power available to process thereceived signal; and allocating processing power to the received signalbased on the availability of processing power.
 11. The method of claim10 further comprising the steps of: determining whether each of aplurality of received signals has human voice content; and assigningprocessing resources to each of the plurality of received signals basedon whether each has human voice content.
 12. The method of claim 11further comprising the steps of: determining a plurality of estimationsof processing resources needed by individual terminals of the pluralityof terminals, wherein the individual terminals are engaged incommunications sessions, allocating processing resources to theindividual terminals based on the plurality of estimations.
 13. Themethod of claim 1 wherein the microphone and the not more than oneactive speaker are proximate to each other and fixably attached to theterminal.
 14. The method of claim 1, wherein the processing stepcomprises the step of: performing acoustic echo cancellation on thereceived signal.
 15. The method of claim 1, wherein the processing stepcomprises the step of: performing automatic gain control on the receivedsignal.
 16. The method of claim 1, wherein the processing step comprisesthe step of: performing noise reduction on the received signal.
 17. Themethod of claim 1, wherein the transmitting step comprises the steps of:sending a time tag, wherein the time tag includes sequencing informationfor the mixed signal.
 18. The method of claim 1 further comprising thestep of: receiving a time tag with the audio signal, wherein theprocessing step aligns the audio signal responsive to the received timetag.
 19. A communications system comprising: a processing switch coupledto a plurality of terminals, wherein each of the plurality of terminalshas a microphone and a speaker proximate to the microphone, wherein theprocessing switch receives a first audio signal from a first terminal ofthe plurality of terminals resulting in a first received audio signal,wherein upon receipt, the first received audio has not been processed toreduce acoustic feedback; a communications support module (CSM) coupledto the processing switch; a signal processing (SP) module coupled to theCSM, wherein the SP module has digital signal processing resources toreduce acoustic feedback in the first received audio signal to result ina first processed signal, wherein the SP module operates under thecontrol of the CSM, a room module coupled to the SP module, wherein theroom module accesses a plurality of room models, wherein each room modelcontains acoustic information specific to one of the plurality ofterminals, wherein each room model contains data related to the acousticproperties surrounding its respective terminal; a mixing module coupledto the processing switch and directed by the CSM, wherein the mixingmodule mixes the first processed signal with a second processed signalto produce an output mix, wherein the output mix is associated with thefirst terminal; and a module for sending the output mix to the firstterminal for output on the first terminal's speaker, wherein the firstterminal has less than two speakers active simultaneously.
 20. Thecommunications system of claim 19, wherein the SP module performsacoustic echo cancellation on the first received audio signal.
 21. Thecommunications system of claim 19, wherein the SP module performsautomatic gain control on the first received audio signal.
 22. Thecommunications system of claim 19, wherein the SP module performs noisereduction on the first received audio signal.
 23. The communicationssystem of claim 19 further comprising: a decoding module for decodingthe first received audio signal; and an encoding module for encoding theoutput mix prior to the output mix being sent to the first terminal. 24.The communications system of claim 19, wherein the CSM determines,responsive to a load on the communications system, an amount of SPmodule processing resources to allocate to individual audio signalswithin a plurality of received audio signals.
 25. The communicationssystem of claim 19, wherein the CSM determines, responsive tocharacteristics of each of a plurality of received audio signals, anamount of SP processing resources to allocate to individual audiosignals within a plurality of audio signals.
 26. The communicationssystem of claim 23, wherein the decoding module is adapted for decodingat least one encoded output mix to produce a decoded encoded output mix,wherein the decoded encoded output mix is used by the SP module toperform acoustic echo cancellation.
 27. The communications system ofclaim 23, wherein the room module updates a first room model based ondata extracted from the first received audio signal, wherein the data isindicative of acoustic characteristics of an area surrounding the firstterminal.
 28. A system for supporting a communications session among aplurality of terminals, the system comprising: a communications supportmodule (CSM) for determining whether a first terminal of the pluralityof terminals is active, wherein responsive to determining that the firstterminal is active, the CSM allocates pooled feedback-reducing resourcesto the first terminal, wherein responsive to determining that the firstterminal is not active, the CSM deallocates pooled feedback-reducingresources from the first terminal; a receiving module for receiving afirst audio signal from the first terminal, wherein the first audiosignal has a feedback component caused by the proximity of a firstterminal's speaker to a first terminal's microphone, wherein the firstterminal has less than two speakers active simultaneously; a room modulefor storing a first room model associated with the first terminal,wherein the first room model contains information relates to acousticproperties influenced by the environment surrounding the firstterminal's microphone and speaker; and a centralized signal processing(SP) module for performing SP on the first audio signal to compensatefor the feedback component, wherein performing SP on the first audiosignal results in a first processed signal, the wherein performing SP onthe first audio signal is responsive to the first room model; and asending module for sending the first processed signal to the firstterminal.
 29. The system of claim 28, further comprising: a mixingmodule for mixing the first processed signal with a second processedsignal to result in a first mixed signal, wherein the sending modulesends the first mixed signal to the first terminal.
 30. The system ofclaim 28, wherein the SP module is adapted to perform acoustic echocancellation on the first audio signal.
 31. The system of claim 28,wherein the SP module is adapted to perform automatic gain control onthe first audio signal.
 32. The system of claim 28, wherein the SPmodule is adapted to perform noise reduction on the first audio signal.33. The system of claim 29, wherein the first audio signal is encoded,the system further comprising: an encoding/decoding module for decodingthe first audio signal and for encoding the first mixed signal.
 34. Thesystem of claim 28, wherein the SP resources are allocated to the firstaudio signal responsive to a total number of available SP resources. 35.The system of claim 28, wherein the SP resources are allocated to thefirst audio signal responsive to content characteristics of the firstaudio signal.
 36. The method of claim 28, wherein the first audio signalis sent in data packets, wherein determining whether the first terminalis active comprises the steps of: analyzing a plurality of received datapackets to determine whether the plurality of data packets originatefrom the first terminal.